Shown in FIG. 1 is a simplified telecommunications network, generally indicated by the numeral 100, that illustrates the basic process and network components involved in the placement of a typical voice-type call. Telecommunications network 100 includes both a calling party (CgPA) 102 and a called party (CdPA) 104. Calling party 102 is communicatively coupled to an originating End Office (EO) or Service Switching Point (SSP) 106, while called party 104 is similarly connected to a terminating EO or SSP 108. Originating SSP 106 and terminating SSP 108 are, in turn, connected via voice-grade communication trunks or links to a tandem switching office 110. SSP 106 and SSP 108 are also connected via signaling links to a Signal Transfer Point (STP) 112. Those skilled in the art of telecommunication network design and operation will appreciate that a typical call setup process begins when calling party 102 goes off-hook and begins dialing a telephone number associated with the called party 104. As such, originating SSP 106 receives and interprets the digits dialed by calling party 102 and subsequently selects one of a plurality of voice-grade links for use with the attempted call. Having selected and reserved a specific voice grade link, SSP 106 then formulates an Integrated Service Digital Network (ISDN) User Part (ISUP) Initial Address Message (IAM) that is intended, at least in part, to communicate or coordinate voice-grade link selection with the tandem switching office 110. Such an ISUP IAM message is typically transported via a Signaling System 7 (SS7) signaling link to STP 112. STP 112 receives the message, examines the message routing label or address header information contained therein, and simply routes the message to the specified destination address which, in this case, corresponds to tandem switching office 110. Using the voice-grade link selection information contained within the ISUP IAM message, tandem switching office 110 is able to reserve the specified link, and consequently a voice-grade communication path is established between SSP 106 and tandem switching office 110. In a similar manner, ISUP messages are transmitted and received by tandem switching office 110, STP 112, and SSP 108 such that a voice-grade communication path is also established between tandem switching office 110 and the terminating SSP 108. Once all of the necessary voice-grade links have been acquired and placed in service, call setup is considered complete and calling party 102 is able to engage in speech-type communication with the called party 104.
While the simplified network shown in FIG. 1 is indicative of the function and service traditionally provided by the Public Switched Telephone Network (PSTN), the present network increasingly carries data, including communications to and from the Internet or World Wide Web (WWW). Furthermore, as the overall performance and reliability of the Internet has improved, so has the incentive to make use of the Internet for the purposes of communicating voice-type calls.
FIG. 2 illustrates one of the most common call flow pathways associated with such “Internet calls”, as described above. More particularly, FIG. 2 includes a communication network generally indicated by the numeral 150. Network 150 is further comprised of both a calling party 152 and a called party ISP 158. Calling party 152 is communicatively coupled to the originating EO or SSP 106, while called party ISP 158 is similarly connected to the terminating EO or SSP 108. Originating SSP 106 and terminating SSP 108 are, in turn, connected via voice-grade communication trunks or links 154 and 156, respectively, to a tandem switching office 110. SSP 106, SSP 108 and tandem switching office 110 are also connected via signaling-grade links to the STP 112.
As previously discussed, call setup is effected between the involved network elements through the use of an appropriate sequence of SS7 signaling messages. In the example shown in FIG. 2, it will be appreciated that although the call is not voice-related the communication links that are allocated and effectively comprise the call pathway are voice-grade links 154 and 156. More particularly, in order for calling party 152 to obtain access to the data network 160, a telephony service provider must employ or utilize some portion of their available voice-grade trunking resources. While functional, such a call scenario is unattractive to telephony service providers for a number of reasons. The two most significant reasons being that expensive, voice-grade trunks are being monopolized to carry data-grade traffic that could otherwise be transported on less expensive data-grade trunks, and that such a scenario creates “convergence” problems at the terminating end office facility, SSP 108. With particular regard to the “convergence” phenomena, it will be appreciated that at any given time, a plurality of calls to an ISP could be placed by a plurality of calling parties where each calling party is serviced by a different originating End Office or SSP. As such, it is possible that the volume of calls facilitated by any individual originating SSP is relatively light. However, it will be appreciated that the terminating SSP which is servicing the called party ISP is required to simultaneously handle or make available sufficient voice-grade trunking to accommodate all of the calls placed by the calling parties. As such, call related traffic is said to “converge” at the terminating SSP that is servicing the called party ISP. Thus, in general, the more Internet service subscribers an ISP is able to recruit, the more severe the terminating SSP or EO convergence problem.
Consequently, there is a significant incentive for telephony service providers to implement new network architectures and equipment that enable both non-voice and voice type calls to be connected or completed via data-grade trunking as opposed to traditional voice-grade trunking. With particular regard to the problem of transporting voice-type traffic through a data network, it will be appreciated that the network architecture illustrated in FIG. 3 has been previously proposed and implemented to provide such “voice over IP” call functionality.
Shown in FIG. 3 is a communications network generally indicated by the numeral 180 which includes components of traditional PSTN type networks as well as traditional data networks such as the Internet 160. Furthermore, network 180 includes a collection of inter-networking elements intended to facilitate communication between the PSTN and data network 160. More specifically, network 180 includes a calling party terminal 102, and a called party terminal 104. Calling party 102 is communicatively coupled to an originating SSP 106, and in a similar manner, called party 104 is communicatively coupled to a terminating SSP 108. SSPs 106 and 108 are in turn connected to an STP 112 via SS7 signaling links. Those skilled in the art of telephony communications will appreciate that such components are typically incorporated within a traditional PSTN type network.
Also coupled to STP 112 are a pair of Media Gateway Controller (MGC) nodes 182 and 184. The MGC nodes provide inter-connectivity and inter-networking functionality between PSTN type network components and data network 160. More particularly, MGC 182 is assigned a unique SS7 Point Code (PC) of 1-1-2 and is connected to STP 112 via a dedicated SS7 signaling link. In a similar manner, MGC 184 is assigned a unique PC of 1-1-3 and is coupled to STP 112 via an SS7 communication link. As such, MGC 182 and MGC 184 are adapted to receive, process and respond to SS7 call setup/teardown signaling messages. Further coupled to MGCs 182 and 184 via signaling links are Media Gateways (MGs) 186 and 188, respectively. It will be appreciated from FIG. 3 that each MG element includes at least three communication interfaces. More specifically, MG 186 is adapted to communicate via a data-grade trunk with SSP 106. MG 186 is also adapted to communicate via a signaling link with MGC 182, while communicating via a data-grade link with data network 160. In a similar manner MG 188 is coupled to SSP 108 via a data-grade trunk, to MGC 184 via a signaling link, and to data network 160 via a data-grade link.
As such, MGC 182 Is able to signal MG 186 in a manner so as to cause MG 186 to establish a data-grade trunk connection with SSP 106, thereby providing a SSP 106 with access to data network 160 without requiring the use of any voice-grade circuit or trunk resources. In a similar manner, MGC 184 and MG 188 provide SSP 108 with the same benefits.
It will be appreciated that in a less optimized configuration, the communication trunking between the SSPs (106 and 108) and the MGs (186 and 188) could be voice-grade. While such a configuration constitutes a less optimized solution than an all data-grade trunk pathway, benefits may still be realized by eliminating the use of tandem office connected voice-grade trunks in scenarios that would ordinarily require multiple tandem offices to be involved in the completion of a call.
From an operational perspective, it should be noted that in practice, both data and voice trunks connected to an SSP or End Office are actually comprised of multiple communication channels or pathways which are commonly referred to as communication circuits. Within any given trunk, these individual communication circuits are identified by a parameter known as a Circuit Identification Code (CIC).
In the example shown in FIG. 3, the establishment of a “call” involves the selection of a particular circuit in a trunk that directly or indirectly facilitates connection of the calling or originating SSP 106 and called or terminating SSP 108. For example, if subscriber 102 that Is serviced by SSP 106 wishes to place a call to another subscriber 104 that is serviced by SSP 108, an SS7 signaling message is formulated and sent from SSP 106 via STP 112 to MGC 182. More specifically, an ISUP IAM message is formulated by SSP 106 indicating that a particular trunk circuit has been selected and reserved for use with the requested call. Within the IAM message, the chosen trunk circuit is indicated by a CIC parameter. The SS7 ISUP IAM message is addressed to the unique SS7 point code associated with MGC 182, which in this example is 1-1-2.
It should be appreciated that the STP 112 simply receives the ISUP IAM signaling message from SSP 106 and routes the message out the appropriate signaling link to MGC 182 based on the Destination Point Code (DPC) specified In the message. Once again, In this example, the DPC of the ISUP IAM message is 1-1-2.
In general, MGC 182 receives the ISUP IAM message and examines the CIC parameter. Based on the CIC value included in the SS7 signaling message, MGC 182 subsequently signals the MG node that is adapted to communicate with SSP 106 via the specified trunk circuit. In this example, the ISUP IAM message is assumed to specify a CIC value that is representative of a trunk circuit maintained by MG 186. Consequently, after receiving the ISUP IAM message, MGC 182 further sends a signaling message to MG 186 so as to generally instruct MG 186 to reserve the trunk circuit requested by SSP 106.
In a similar manner, SS7 ISUP messages are also between MGC 182, MGC 184, and terminating SSP 108 so as to effectively establish a call pathway between the calling party 102 and called party 104. In this case, the calling pathway includes, at least in part, a data network component and furthermore does not require an Internet Service Provider (ISP) to provide access to this data network component.
While the network architecture described above offers numerous benefits over previous “Internet call” processing implementations, one significant limitation of such an architecture involves the requirement that each MGC node be assigned a unique SS7 network address or point code (PC). With the rapid expansion of the PSTN, SS7 point codes have become a scarce resource. Consequently, it is not always feasible for a telephone network operator to implement new network architectures or network growth plans that require the acquisition of numerous new SS7 point codes.
Therefore, what is needed is a system and method of establishing calls, at least in part, through a data network using inter-networking nodes that do not require unique SS7 point codes.